Debugging VOIP with Shikha Sharma #CiscoKid

========================================================================== Frame Relay router ! interface Serial0/0/0  description *** static DCE connection to s1 Prasit  no ip address  encapsulation frame-relay  clock rate 2000000  frame-relay intf-type dce  frame-relay route 150 interface Serial0/1/0 130  frame-relay route 170 interface Serial0/2/0 180 ! interface Serial0/1/0  description *** static DCE connection to s0 spicy  bandwidth 1000000  no ip address  encapsulation frame-relay  tx-ring-limit 100  tx-queue-limit 100  transmitter-delay 10  frame-relay intf-type dce  frame-relay route 130 interface Serial0/0/0 150  frame-relay route 140 interface Serial0/2/0 160 ! interface Serial0/2/0  description *** static DCE connection to s1 Aton  ip address 192.168.22.1 255.255.255.0  encapsulation frame-relay  frame-relay intf-type dce  frame-relay route 160 interface Serial0/1/0 140  frame-relay route 180 interface Serial0/0/0 170 ! ================================================

Building configuration... Current configuration : 3093 bytes ! version 12.4 service timestamps debug datetime msec service timestamps log datetime msec no service password-encryption ! hostname PSTN-CME-VG ! boot-start-marker boot-end-marker ! no aaa new-model clock timezone India 5 30 network-clock-participate slot 1 voice-card 0  no dspfarm ! voice-card 1  dspfarm ! voice-card 2  dspfarm ! dot11 syslog no ip dhcp use vrf connected ip dhcp excluded-address 192.168.15.1 192.168.15.10 ! ip dhcp pool PSTN-Phones    network 192.168.15.0 255.255.255.0    default-router 192.168.15.2    option 150 ip 192.168.15.2 ! ! ip cef ! ! multilink bundle-name authenticated ! isdn switch-type primary-ni ! ! ! ! ! ! ! ! ! ! ! ! ! ! ! ! ! ! ! ! archive  log config   hidekeys ! ! controller T1 1/0/0  framing esf  clock source internal  linecode b8zs  pri-group timeslots 1-12,24 ! controller T1 1/0/1  framing esf  linecode b8zs ! controller T1 2/0/0  framing esf  linecode b8zs ! ip tftp source-interface G

Fractional PRI with MGCP is not supported. But there is a workaround. To avoid any problems you can busy out channels that are not in use on the Gateway from the ccmadmin pages. This way Callmanager will not try to use channels that are unavailable. This is a workaround. Here is the bug ID- CSCdw78580 From CCMAdmin pages go to System > Service Parameters > Service Parameter Configuration > Clusterwide Parameters (Device - PRI and MGCP Gateway) -> Change B-Channel Maintenance Status 1 B-Channel maintenance status for PRI and CAS interfaces in real time for  troubleshooting. Input Format: Device Name = B-channel Maintenance Status Example: S0/DS1-0@SDA123456789ABC = 0000 1111 0000 0000 0000 0001 (24 bits for  T1. Channel number begins with 1 to 24 from left to right.* In the case of PRI, the last bit specifies the D Channel*, which does not get  affected. *In this example, the 5th through 8th B-Channels are out of service.*) or, S1/DS1-0@DLS2-CM136-VG200-2.SELSIUSCMLAB.CI

Essentially only three steps are required to configure MGCP gateway. R1(config)#mgcp R1(config)#ccm-manager config R1(config)#ccm-manager mgcp server <Primary TFTP IP address> R1(config)#ccm-manager redundant-host <Secondary TFTP IP address> --- this command is optional, as suggests it provides secondary TFTP server as a redundant host. A couple of important points to keep in mind : 1. Configure domain name on the gateway and configure gateway configuration on CUCM with the correct domain name. 2. Bind media with the source interface- R1(config)#mgcp bind control source-interface <> R1(config)#mgcp bind media source-interface <> 3. As soon as you configure pri-group timeslots with service mgcp under controller, it'll b ind Lthe ayer 3 of the ISDN PRI interface of the MGCP voice gateway to the CUCM for PRI Q.931 signaling backhaul support interface Serial0/0/0:23 isdn bind-l3 ccm-manager

#service-module service-Engine 1/0 session Trying 172.31.1.10, 2066 ... % Connection refused by remote host #service-module service-Engine 1/0 status Service Module is Cisco Service-Engine1/0 Service Module supports session via TTY line 66 Service Module is in Steady state Service Module heartbeat-reset is enabled Getting status from the Service Module, please wait.. Cisco Unity Express 7.0.3 CUE Running on NM   Check your line 66 configuration, in my case, I had to change the transport input to telnet- line 66  no activation-character  no exec  transport preferred none  transport input telnet  transport output telnet

The error I am referring here is- Error: Unable to login as administrator to the Cisco CallManagerExpress at site 'local'. Details: CME login failed To fix this issue, confirm the following steps- 1. Under telephony-service, provide the correct username and password of CME. 2. Enable "ip http server" on CME. 3. On CUE, check "CallManager Express" configuration, CME IP address should be correct as well as SIP gateway address should be CME IP. 4. Try synchronizing the database with CME.

When the softphone starts up it requests configuration file in the following order: VTGOPC.xml VTGO.cfg.xml VTGOPC.xml VTGO.cfg.xml SEP000000000001.cnf.xml (Repeating) You can easily create the VTGOPC.xml file as follows CME#conf t Enter configuration commands, one per line.  End with CNTL/Z. CME(config)#telephony-service CME(config-telephony)#create cnf-files CNF file creation is already On Creating CNF files CME(config-telephony)#do show telephony tftp-bindings tftp-server system:/its/united_states/7960-tones.xml alias United_States/7960-tones.xml tftp-server system:/its/united_states/7960-font.xml alias English_United_States/7960-font.xml tftp-server system:/its/united_states/7960-font.xml alias English_United_States/7920-font.xml tftp-server system:/its/united_states/7960-dictionary.xml alias English_United_States/7960-dictionary.xml tftp-server system:/its/united_states/7960-kate.xml alias English_United_States/7960-kate.xml tftp-server system:/its/united_states/7960-kate

1. Define bearer-cap speech in your voice-port to get this sorted. 2. If the first option doesn't apply in your scenario, check the codec. Its a clear issue of codec mismatch.

Resolution- Restart DRF Master service. For me, it didn't work instantly but suddenly the next day it started working.

Simple solution- Open Command Line and run the following commands, this will shrink your unity DB- osql -E 1> use UnityDB 2> go 1> backup log UnityDB with no_log 2> go 1> dbcc shrinkdatabase(UnityDB) 2> go 1>exit osql -E 1> use ReportDB 2> go 1> backup log ReportDB with no_log 2> go 1> dbcc shrinkdatabase(ReportDB) 2> go 1>exit

Sample call flow of such scenario- IP Phone CME A >> IP Phone CME B >> Call forward to another extension of CME B. In call forward loop scenario, Call will reroute by CME A instead of CME B. This scenario gets complicated when we call forward the calls to a PSTN number, in such scenarios, call routes via CME A, still doesn't looked complicated, let me give you an example- India IP phone calling directly to US IP phone which is CFA to local mobile number which could be 9.1234567890 When India IP phone hits the US CME, it will send the call forward information to India CME which is 9.1234567890, India CME won;t able to complete this call because it is not its local number rather international. To avoid this call forward looping, you'd need to disable H450.3 service in the forwarding CME. It is activated by default on a CME with 3.1 or later versions, this service loops the call. The idea of this service is to minimize the resources being used at re

1. Configure night-service under telephony-system- night-service day Sun 08:00 07:59 night-service day Sat 08:00 07:59 night-service weekday 16:00 08:00 The time will comprise - from - when you want night-service to invoke - to- when you want it to come down. 2. Apply night-service on the DN- ephone-dn  100 night-service bell call-forward night-service 5000 number 1000 Point to be noted- Make sure the time on CME and time on the phone matches, if it doesn't check this post .